Consider SIP Protocol for Voice Over IP
SIP (Session Initiation Protocol) is the transport protocol for person-to-person voice traffic over the Internet. Its defining specifications come from the SIP working group of the Internet Engineering Task Force. SIP gives access to the public switched telephone network for voice data at 3 kHz bandwidth and common number dialing using Voice over IP (VoIP). It can also be extended to handle IP telephony in conjunction with video and instant messaging. Future enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Due West.
SIP is commonly associated with a method to implement the operation of typical telephony over an IP network. It is replacing the older, less flexible protocols used in the past such as H.323 and MGCP. These earlier protocols worked at a very low level to connect IP phones to the public telephone system. SIP, however, provides an elegant and well-defined way to network the enterprise. For example, SIP uses email addresses as the SIP address in place of a telephone number over the standard phone network.
SIP trunking provides multiple perks to the business user such as cost savings, networking flexibility, and emergency disaster recovery. It can do away with the high subscription expenses of basic rate interfaces and primary rate interfaces. It also optimizes the bandwidth usage by delivering both voice and data over the very same connection. Businesses will have the manageability to route calls to favored carriers and the redundancy of using a number of service providers.
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